webrtc_source_file: remove elements created when error occurred

[Version] 0.3.184
[Issue Type] Improvement

Change-Id: I6fc4c73e27a031c122abfba77b10d8cd8b0064ee

diff --git a/packaging/capi-media-webrtc.spec b/packaging/capi-media-webrtc.spec
index 7bc23cd6ab3b0bf6542bd52d7d5e9eef3c763334..2b8b3ab6f329e9cb8de31b07b4296d89dda5cd7f 100644 (file)
--- a/packaging/capi-media-webrtc.spec
+++ b/packaging/capi-media-webrtc.spec
@@ -1,6 +1,6 @@
 Name:       capi-media-webrtc
 Summary:    A WebRTC library in Tizen Native API
-Version:    0.3.183
+Version:    0.3.184
 Release:    0
 Group:      Multimedia/API
 License:    Apache-2.0

diff --git a/src/webrtc_source_file.c b/src/webrtc_source_file.c
index 175f827cca38959cd9228f256593788cc9c8d1d3..471cb4d1c8eb4e7407d105065c820577aa4a697d 100644 (file)
--- a/src/webrtc_source_file.c
+++ b/src/webrtc_source_file.c
@@ -281,14 +281,14 @@ static GstElement * __prepare_fakesink_for_filesrc_pipeline(webrtc_gst_slot_s *s
        return fakesink;
 }
 
-static int __create_rest_of_elements_for_filesrc_pipeline(webrtc_gst_slot_s *source, GstPad *pad, bool is_audio)
+static int __create_rest_of_elements_for_filesrc_pipeline(webrtc_gst_slot_s *source, GstPad *pad, GList **element_list, bool is_audio)
 {
        GstBin *bin;
        GstElement *queue;
        GstElement *payloader;
        GstElement *capsfilter;
        GstElement *fakesink;
-       GList *element_list = NULL;
+       GList *_element_list = NULL;
 
        RET_VAL_IF(source == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "source is NULL");
        RET_VAL_IF(source->filesrc_pipeline == NULL, WEBRTC_ERROR_INVALID_OPERATION, "filesrc_pipeline is NULL");
@@ -297,41 +297,41 @@ static int __create_rest_of_elements_for_filesrc_pipeline(webrtc_gst_slot_s *sou
 
        if (!(queue = _create_element(DEFAULT_ELEMENT_QUEUE, _get_element_name(GET_AV_IDX(is_audio), ELEMENT_QUEUE))))
                return WEBRTC_ERROR_INVALID_OPERATION;
-       APPEND_ELEMENT(element_list, queue);
+       APPEND_ELEMENT(_element_list, queue);
 
        if (!(payloader = __create_payloader_for_filesrc_pipeline(pad, is_audio)))
                goto exit;
-       APPEND_ELEMENT(element_list, payloader);
+       APPEND_ELEMENT(_element_list, payloader);
 
        if (!(capsfilter = __prepare_capsfilter_for_filesrc_pipeline(source, is_audio)))
                goto exit;
-       APPEND_ELEMENT(element_list, capsfilter);
+       APPEND_ELEMENT(_element_list, capsfilter);
 
        if (!(fakesink = __prepare_fakesink_for_filesrc_pipeline(source, is_audio)))
                goto exit;
-       APPEND_ELEMENT(element_list, fakesink);
+       APPEND_ELEMENT(_element_list, fakesink);
 
-       if (!_add_elements_to_bin(bin, element_list)) {
-               SAFE_G_LIST_FREE(element_list);
+       if (!_add_elements_to_bin(bin, _element_list)) {
+               SAFE_G_LIST_FREE(_element_list);
                return WEBRTC_ERROR_INVALID_OPERATION;
        }
 
-       if (!_link_elements(element_list))
+       if (!_link_elements(_element_list))
                goto exit_with_remove_from_bin;
 
-       if (!_sync_elements_state_with_parent(element_list))
+       if (!_sync_elements_state_with_parent(_element_list))
                goto exit_with_remove_from_bin;
 
-       SAFE_G_LIST_FREE(element_list);
+       *element_list = _element_list;
 
        return WEBRTC_ERROR_NONE;
 
 exit_with_remove_from_bin:
-       _remove_elements_from_bin(bin, element_list);
-       SAFE_G_LIST_FREE(element_list);
+       _remove_elements_from_bin(bin, _element_list);
+       SAFE_G_LIST_FREE(_element_list);
        return WEBRTC_ERROR_INVALID_OPERATION;
 exit:
-       SAFE_G_LIST_FREE_FULL(element_list, gst_object_unref);
+       SAFE_G_LIST_FREE_FULL(_element_list, gst_object_unref);
        return WEBRTC_ERROR_INVALID_OPERATION;
 }
 
@@ -452,14 +452,14 @@ static GstAutoplugSelectResult __filesrc_pipeline_decodebin_autoplug_select_cb(G
        return GST_AUTOPLUG_SELECT_TRY;
 }
 
-static int __build_filesrc_bin(webrtc_gst_slot_s *source, media_type_e media_type)
+static int __build_filesrc_bin(webrtc_gst_slot_s *source, media_type_e media_type, GList **element_list)
 {
        int ret = WEBRTC_ERROR_NONE;
        GstPad *src_pad = NULL;
        GstElement *appsrc = NULL;
        GstElement *queue = NULL;
        GstElement *capsfilter = NULL;
-       GList *element_list = NULL;
+       GList *_element_list = NULL;
        const int av_idx = GET_AV_IDX_BY_TYPE(media_type);
 
        RET_VAL_IF(source == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "source is NULL");
@@ -472,7 +472,7 @@ static int __build_filesrc_bin(webrtc_gst_slot_s *source, media_type_e media_typ
 
        if (!(appsrc = _create_element(DEFAULT_ELEMENT_APPSRC, _get_element_name(av_idx, ELEMENT_APPSRC))))
                return WEBRTC_ERROR_INVALID_OPERATION;
-       APPEND_ELEMENT(element_list, appsrc);
+       APPEND_ELEMENT(_element_list, appsrc);
 
        g_object_set(G_OBJECT(appsrc),
                "is-live", TRUE,
@@ -481,21 +481,21 @@ static int __build_filesrc_bin(webrtc_gst_slot_s *source, media_type_e media_typ
 
        if (!(queue = _create_element(DEFAULT_ELEMENT_QUEUE, _get_element_name(av_idx, ELEMENT_QUEUE))))
                goto exit;
-       APPEND_ELEMENT(element_list, queue);
+       APPEND_ELEMENT(_element_list, queue);
 
        if (!(capsfilter = _create_element(DEFAULT_ELEMENT_CAPSFILTER, _get_element_name(av_idx, ELEMENT_CAPSFILTER))))
                goto exit;
-       APPEND_ELEMENT(element_list, capsfilter);
+       APPEND_ELEMENT(_element_list, capsfilter);
 
-       if (!_add_elements_to_bin(source->bin, element_list)) {
-               SAFE_G_LIST_FREE(element_list);
+       if (!_add_elements_to_bin(source->bin, _element_list)) {
+               SAFE_G_LIST_FREE(_element_list);
                return WEBRTC_ERROR_INVALID_OPERATION;
        }
 
-       if (!_link_elements(element_list))
+       if (!_link_elements(_element_list))
                goto exit_with_remove_from_bin;
 
-       if (!_sync_elements_state_with_parent(element_list))
+       if (!_sync_elements_state_with_parent(_element_list))
                goto exit_with_remove_from_bin;
 
        if (_set_ghost_pad_target(src_pad, capsfilter, true) != WEBRTC_ERROR_NONE)
@@ -503,16 +503,16 @@ static int __build_filesrc_bin(webrtc_gst_slot_s *source, media_type_e media_typ
 
        _add_probe_to_pad_for_pause(source, av_idx, src_pad, _payloaded_data_probe_cb);
 
-       SAFE_G_LIST_FREE(element_list);
+       *element_list = _element_list;
 
        return WEBRTC_ERROR_NONE;
 
 exit_with_remove_from_bin:
-       _remove_elements_from_bin(source->bin, element_list);
-       SAFE_G_LIST_FREE(element_list);
+       _remove_elements_from_bin(source->bin, _element_list);
+       SAFE_G_LIST_FREE(_element_list);
        return WEBRTC_ERROR_INVALID_OPERATION;
 exit:
-       SAFE_G_LIST_FREE_FULL(element_list, gst_object_unref);
+       SAFE_G_LIST_FREE_FULL(_element_list, gst_object_unref);
        return WEBRTC_ERROR_INVALID_OPERATION;
 }
 
@@ -556,6 +556,8 @@ static void __filesrc_pipeline_decodebin_pad_added_cb(GstElement *element, GstPa
        gchar *media_type = NULL;
        bool is_audio;
        int av_idx;
+       GList *filesrc_element_list = NULL;
+       GList *bin_element_list = NULL;
        GstElement *queue = NULL;
 
        RET_IF(source == NULL, "source is NULL");
@@ -584,14 +586,17 @@ static void __filesrc_pipeline_decodebin_pad_added_cb(GstElement *element, GstPa
                return;
        }
 
-       ret = __create_rest_of_elements_for_filesrc_pipeline(source, pad, is_audio);
+       ret = __create_rest_of_elements_for_filesrc_pipeline(source, pad, &filesrc_element_list, is_audio);
        if (ret != WEBRTC_ERROR_NONE) {
                LOG_ERROR("failed to __create_rest_of_elements_for_filesrc_pipeline()");
                return;
        }
 
        queue = gst_bin_get_by_name(GST_BIN(source->filesrc_pipeline), _get_element_name(GET_AV_IDX(is_audio), ELEMENT_QUEUE));
-       RET_IF(queue == NULL, "queue is NULL");
+       if (!queue) {
+               LOG_ERROR("failed to get queue");
+               goto exit;
+       }
 
        ret = __link_decodebin_with_queue(pad, source, is_audio);
        if (ret != WEBRTC_ERROR_NONE) {
@@ -599,7 +604,7 @@ static void __filesrc_pipeline_decodebin_pad_added_cb(GstElement *element, GstPa
                goto exit;
        }
 
-       ret = __build_filesrc_bin(source, is_audio ? MEDIA_TYPE_AUDIO : MEDIA_TYPE_VIDEO);
+       ret = __build_filesrc_bin(source, is_audio ? MEDIA_TYPE_AUDIO : MEDIA_TYPE_VIDEO, &bin_element_list);
        if (ret != WEBRTC_ERROR_NONE) {
                LOG_ERROR("failed to __build_filesrc_bin()");
                goto exit;
@@ -608,20 +613,30 @@ static void __filesrc_pipeline_decodebin_pad_added_cb(GstElement *element, GstPa
        ret = _link_source_with_webrtcbin(source, source->webrtc->gst.webrtcbin);
        if (ret != WEBRTC_ERROR_NONE) {
                LOG_ERROR("failed to _link_source_with_webrtcbin()");
-               goto exit;
+               goto exit_with_remove_from_bin;
        }
 
        source->av[av_idx].render.need_decoding = true;
        source->av[av_idx].render.appsrc_caps = gst_pad_get_current_caps(pad);
        _add_probe_to_pad_for_render(source, av_idx, gst_element_get_static_pad(queue, "src"), _source_data_probe_cb);
 
+       SAFE_G_LIST_FREE(filesrc_element_list);
+       SAFE_G_LIST_FREE(bin_element_list);
+
        GENERATE_DOT(source->webrtc, source->filesrc_pipeline, "%s.%s-%s",
                GST_ELEMENT_NAME(source->filesrc_pipeline), GST_ELEMENT_NAME(element), GST_PAD_NAME(pad));
 
+       GENERATE_DOT(source->webrtc, source->webrtc->gst.pipeline, "%s.filesrc.%s-%s",
+               GST_ELEMENT_NAME(source->webrtc->gst.pipeline), GST_ELEMENT_NAME(element), GST_PAD_NAME(pad));
+
        return;
 
+exit_with_remove_from_bin:
+       _remove_elements_from_bin(GST_BIN(source->bin), bin_element_list);
+       SAFE_G_LIST_FREE(bin_element_list);
 exit:
-       _remove_rest_of_elements_for_filesrc_pipeline(source, is_audio);
+       _remove_elements_from_bin(GST_BIN(source->filesrc_pipeline), filesrc_element_list);
+       SAFE_G_LIST_FREE(filesrc_element_list);
        return;
 }